1. Field of the Invention
The invention relates a relay device for relaying data in a way that mutually converts voice signals and IP packets between an existing telephone network and an IP (Internet Protocol) network.
2. Description of the Related Art
At the present, utilization of a technology for performing a voice talk is underway, wherein the existing Public Switched Telephone Networks (which will hereinafter be abbreviated to PSTNs) are connected to each other via an IP (Internet Protocol) network. This type of technology involves employing VOIP (Voice Over Internet Protocol) etc. for transferring and receiving the voice signals over the IP network.
This type of VOIP-based conventional voice talk system (which will hereinafter be referred to as a conventional system) will be explained with reference to FIG. 17. FIG. 17 is a view showing an example of a network architecture of the conventional telephone system. In the conventional system shown in FIG. 17, a VOIP gateway (which will hereinafter be abbreviated to VOIPGW) located at a boundary between the PSTN and the IP network voice-packetizes digital signals (STM (synchronous Transport Module)-1, STM-4, etc.) transferred and received over the PSTN by use of a self-equipped CODEC (Coder/Decoder) etc. and forwards the voice packets to the IP network, thereby actualizing voice communications.
The conventional system is that, as illustrated in FIG. 17, telephones 215 as subscriber terminals connected to a PSTN 211 are connected to telephones 216, etc. connected to another PSTN 212 via an IP network 210. Further, the PSTNs 211 and 212 are connected to the IP network 210 via VOIP gateways 213 and 214, respectively. Moreover, a call agent (which will hereinafter abbreviated to CA) is connected to the IP network 210, wherein this CA controls calls from the respective telephones 215 and 216. Further, an FTP (File Transfer Protocol) server 218 is connected to the IP network 210. The FTP server 218 retains digital data (which will hereinafter be referred to as voice source data) etc. into which a guidance message of a talkie etc. is voice-coded by utilizing a μ-LAW 64 kbs PCM (Pulse Coded Modulation (ITU-T G.711) system and so on.
Next, an operation of the conventional system on the occasion of providing a service for flowing the guidance message of the talkie etc. to the telephone as the subscriber terminal, will be explained with reference to FIGS. 17 and 18. FIG. 17 is a view showing a network architecture of the conventional system and also illustrating how the voice source data are transferred from the FTP server. FIG. 18 is a view showing how the voice source data are sent to the PSTN from the VOIPGW in response to an instruction of the CA in the conventional system shown in FIG. 17.
To start with, as preprocessing, the FTP server 218 transfers, as shown in FIG. 17, the voice source data to the VOIP gateways 213 and 214 (S219). Then, the VOIP gateways 213 and 214 receiving the voice source data store memories with the voice source data. Namely, the voice source data related to the message of the talkie etc. are stored on the respective VOIP gateways.
Next, the operation of the conventional system for actually flowing the message to each telephone, will be described. The conventional system sends the message in response to a call from the telephone as the user terminal. In this case, the CA notifies each VOIPGW of call control information such as call setting, a voice source data add instruction, etc. (S221). The VOIPGW notified of the call control information adds the voice source data to a designated timeslot in the STM, thereby sending the voice source data to a target telephone (S222).
An operation of the VOIPGW stored with voice source data and sending the stored voice source data to the target telephone, will be described with reference to FIG. 19. FIG. 19 is a diagram showing a configuration of the VOIPGW in the conventional system and also illustrating how the VOIPGW is stored with the voice source data and sends the voice source data. Note that FIG. 19 shows a functional configuration by exemplifying the VOIPGW 213.
The VOIPGW 213 is constructed of an IP switch unit 231 serving as an interface with the IP network, an STM switch control unit 232 serving as an interface with the PSTN, a control unit 233, a CODEC unit 234, a packet processing unit 235 and a packet buffer 236. The STM switch control unit 232 is further constructed of a voice source data storage memory 237, a voice source data add unit 238, etc.
In the case of storing the voice source data given from the FTP server 218, the VOIPGW 213 receives the voice source data from the IP network 210 and stores the voice source data on the voice source data storage memory 237 within the STM switch control unit 232 via the IP switch unit 231, the packet processing unit 235 and the control unit 233 (a data flow indicated by a dotted line in FIG. 19).
On the other hand, in the case of sending the voice source data to the telephone, the VOIPGW 213 receives a call control signal from the CA. The VOIP gateway 213 receiving the call control signal from the CA instructs the packet processing unit 235, the CODEC unit 234 and the STM switch control unit 232 to perform call setting in accordance with the call control signal (a data flow indicated by one-dotted broken line in FIG. 19). Next, the VOIPGW 213 receives a voice source data add instruction from the CA. Upon receiving the instruction, the control unit 233 instructs the voice source data add unit 238 to send (add) the voice source data into a channel (call) designated in the voice source data add unit 238 (a data flow indicated by a solid line in FIG. 19).
An operation of the voice source data add unit 238 will be explained in greater detail with reference to FIG. 20. FIG. 20 is a diagram showing a detailed functional configuration of the voice source data add unit 238 in the conventional VOIPGW. The voice source data add unit 238 adds, based on the voice source data add instruction given from the control unit 233, the designated voice source data into the designated channel. Further, respective functional units of the voice source data add unit 238, as one frame is transmitted and received at an interval of 125 micro second (μs) when the PSTN employs the STM-1 communication system, execute the following processes within this interval.
A voice source data readout control unit 241, in accordance with the voice source data add instruction given from the control unit 233, on a channel-by-channel basis, calculates a readout address, reads the voice source data from the voice source data storage memory 237, retains the readout data on an add data storage register 234 (channel unit), and updates and retains the readout address on a voice source data readout address storage register 244 (channel unit). Moreover, a voice source data add processing unit 246 reads the add data from the add data storage register 243 and adds the add data in synchronization with a channel-by-channel transmission timing.
As described above, in the conventional system, the voice source data transmitted by the FTP server 218 are stored on the voice source data storage memory 237 of the VOIP gateway 213. Then, in the case of sending the voice source data in accordance with the call given from the subscriber terminal, the VOIP gateway 213 reads the voice source data on the channel-by-channel basis (the channel unit) from the voice source data storage memory 237, and the voice source data add processing unit 246 adds (allocates) the voice source data to a predetermined timeslot of the STM.
Note that the conventional art related to the present invention of the application is disclosed in the following document. The conventional art document is “Japanese Patent Application Laid-Open Publication No.4-239254”.
The voice source data storage/transmission method in the conventional system, however, has the following problems.
First, in the conventional system, the storage of the voice source data involves preparing a dedicated memory such as a ROM (Read Only Memory) etc. in the STM switch control unit 232 within the VOIPGW in order to store the voice source data. The STM switch control unit 232 is normally mounted with only a small-capacity memory. Therefore, it is required that a memory for storing the voice source data be provided for this purpose. A flash ROM suited to accessing on a 1-byte basis is in many cases employed for this dedicated memory. This is because in the case of adding the data into a timeslot corresponding to each channel, the data are required to be added on the 1-byte basis in terms of STM communications standards. Further, in the case of having a necessity of storing plural categories of voice source data, even when using a large-capacity flash ROM, a plurality of memories are needed. For example, the flash ROM having an 8-megabyte (MB) capacity is stored with only the voice source data on the order of 16 min as a total.
Second, on the occasion of adding the voice source data into the STM timeslot, there can be no perfect assurance for searching out the head of the voice message, corresponding to a call of every subscriber terminal. This is derived from the following reasons. In the case of adding the voice source data into the STM timeslot, it is required that the voice source data be separately readout for every channel corresponding to the call. If the number of channels which should be added at a time increases, there must be a rise in data size of the data to be read out within a predetermined frame interval (e.g., 125 μs at 64 Kbps), and hence the memory access speed does not catch up with this rise.
Concerning this problem, there is proposed a method that the voice source data are previously read out at a certain fixed interval in order to search out the head of the voice message, and the closest readout data is selected and added into the target timeslot (refer to “Japanese Patent Application Laid-Open Publication No.4-239254”). In this method also, however, if there are plural categories of voice messages, it follows that a limit of a voice source data readout interval is determined from the memory access time, and hence there is no perfect head-search-out function. For instance, in the case of the voice source data on the order of 16 min as a total, supposing that the access time to the flash ROM is 90 nanoseconds (ns), the readout interval is equal to or larger than approximately 700 milliseconds (ms). In the case of the voice source data on the order of 32 min as a total, the readout interval is equal to or larger than approximately 1.4 sec.
Third, the memory management of the memory (the voice source data storage memory 237) for storing the voice source data of the voice message becomes troublesome. The conventional system, in the case of storing the voice source data corresponding to the voice message, requires ensuring a memory area for a maximum length of the voice source data that should be stored previously. Further, the voice source data, if not stored in one area, are divided into equal data segments and thus stored. Under such a condition, when changing the voice message, especially when changing into a voice message having a different message length, it is required that the memory area already stored with the voice source data be released and that the segmented memories be reallocated to the voice source data for the change.